Zoom 5801

This unit answers a market need for ATA 911 compliance - to a degree. This ATA has a POTS (FXO) connection in addition to the phone connection (FXS). This POTS connection can be engaged by the dialplan set in the ATA, a loss of the VOIP server as well as dialing a special code. So if I deploy these, based on the config file in the ATA, I can set the 911 to use either the POTS line, the VOIP line or the pots as a backup to the voip line and I believe the VOIP line in backup to the POTS line for emergency numbers.

Having an FXO port makes you want to use it for Asterisk! Well it can to a point. You can have the ATA act as a single line SIP FXO port (in and out) or an ATA, but not concurrently - at this time. Read on.


FIRST LETS GET THE ATA TO WORK ON ASTERISK

Using IE (carefull of IEv7 & Firefox - problems reported), open the web interface to the unit. In VOIP Accounts, set up then 'phone number' and 'auth user name' to the extension you wish to assign in Asterisk. Set up the 'Auth Password' the the password for this extension. Next set the Sip server, Outbound Proxy and Register Domain to the Asterisk server's IP address (or FDDN). I will not go into STUN or NAT so testing should be local.

Set up the extension in Asterisk and you should be able to receive calls to the ATA. Placing calls may mean you have to modify the dial plan as well as a few other paramters.

I set the mode to pass all * and # sequences to the Asterisk server (Subscription Services - Dialing Parameters). This is only overridden by the dial plan template. In addition, I had to remove foreign dial plan codes out of my dial template such as 999, 100 and 11x - especially 100 since I was trying to dial extension 100! (Subscription Services - Emergency Services)


PLACING A CALL OUT THE POTS LINE

First make sure you can call the ATA extension phone from Asterisk and call out Asterisk using the extension phone on the ATA.

The Asterisk server can place a call out the POTS line on the unit if it is set up to: Bridge from VOIP to PSTN, Auto answer, accept any call, bridge anonymous call and disable password checking (Subscription Services)

Ceate a CUSTOM Trunk in Asterisk with the String SIP/xxx where xxx is the extension the ata is set up as - exclude the dial string (OUTNUM) since it does nothing.

This does require a custom dial plan since any trunk or outbound route defined in AMP, TrixBox or AAH will not pass the digits to be dialed properly. Basically there are two ways around this. One is custom dial code and the other is to just engage the outbound line immediately.

To engage the outbound line immediately, just define the cutom trunk as above and an outbound route. Make the outbound route have something like 6| without anything else. Also change your phones mac config file to timeout immediately after pressing the 6. Pressing 6 will execute the trunk and cause the 5801 to answer and go off hook on the pots line. You can then dial the number.

The other option is to create custom code so the normal dial system will work. This is what I did.

Since I am using custom code, I can just add the trunk as another trunk in an already configured outbound route - or for testing just create an ordinary route using the Zoom custom trunk. I edited the AMP file extensions.conf which WILL CHANGE on any AMP update. (But not on every AMP configuration change you make as will extensions_additional.conf). I just made a very small change and it is easy to watch out for on AMP updates. Basically, the dial command allows an additional parameter which is not used by AMP or the custom trunks in AMP and that is to dial additional digits after connection. It is the 'D' Trunk Option.

My custom code is to add 'D(w${OUTNUM})' to the end of the custom dial string (just before the last close brace) in extensions.conf as so:

exten => s,n(skipoutnum),Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS}D(w${OUTNUM}))

The w is a delay to wait for the bridge to complete. This is the quickest fix I could find.

Observations about this configuration:
>>Once the bridge is complete and a call is made from the VOIP server, the analogue phone attached to the ATA can pick up and dial an extension on the VOIP server as a regular extension (if the hot dial string is not engaged as stated in my second configuration below)

>>If the phone attached to the ATA is busy on a VOIP call, any attempt to use the ATA as an outbound trunk causes a call waiting beep to be heard on the phone before the bridge is performed.


RECEIVING AN OUTSIDE CALL USING THE POTS LINE

The VOIP server can receive a call as if it was an external SIP FXO adapter if it is set up to: Bridge from PSTN to VOIP, Auto Answer, Accept any call, accept anonymous and disable passwords. In addition, hot line dialing is engaged with a hot dial string (7777 in AMP) to force the server to go into an inbound trunk mode.


This works great except:

>>If there is a phone attached to the unit and it is picked up it also dials the hot line number, making it an outside call (due to the forced dialed digits) instead of an extension. Trying to avoid the hot dial by making a warm dial (user defined delay, currently 4 sec) would work but the same delay is presented to an outside caller. This gives an outside caller a dial tone for 4 seconds and a 'from-internal' context. So forcing the dial is the easiest way without modifying the context or peer type (friend, user, peer) to avoid misuse.

>>When the FXO picks up, the callerid passed to Asterisk is the phone extension rather than the outside caller. The documentation implies the FXO ID is passed, but the SIP packets say otherwise. I'm still checking on this.


MY OPINION

This unit does have a few problems at this time. The sip status always shows the lag at 60ms or greater when it is 5 feet away and on the same switch, but this does not seem to effect echo or performance. A few bugs do exist and the beta version is addressing these. Someone at Zoom has a sence of humor. I got a response to my sip command that stated it was a COFFEE POT not a sip server. :-o) LOL!!!

I tried the beta 4 software for a few hours but I could not, using the same setup, get it to work. Sip status always showed it as LAGGED, making its use impossible. So the beta is not ready yet, but I may have another crack at it.

It is nice to be able to use a single device for both an ATA and an external single port FXO. Less inventory to buy.

Forget about using both channels with Asterisk at the same time FXS & FXO. At this time there is no convenient way. I have emailed Zoom about allowing this. My idea to maintain 911 compliance on the fxs port is when a 911 or pattern match emergency call is being made is to interrupt the FXO port for a period of time forcing the 911 out the pots line. This would revert back to the VOIP server after a preset time or if a special key combination was entered.

In addition, they should stop the local phone from ringing when the bridge from VOIP to PSTN is engaged. Also the beep should be eliminated if the local phone is off hook. However I do realize that my testing is using the second appearance of a single line so I may just have to wait for a new offering from them.

The deployment tools for large installs that allow you to upload new configurations to your clients also need a little more documentation. They state Perl & PHP must be installed, but minimumn versions are not specified. I'm using PHP 4.3 (I Know its a little behind) and I could not run the utilities properly. Also creating the recommended Apache links didn't work, too much depends on the exact structure, so move the whole thing into the html folder or start coding.

Overall, I like it. I'm going to pick up a few more and deploy them in test areas. Now if I could just get a better firmware...



This was based on Rom 1.0.2, Firmware 1.12 & Conf Revision 1.1.1.01

Also confirmed working ROM 1.0.2, Firmware 1.0.7 & Conf Revsion 1.0.8-00SP

Also confirmed working ROM 1.0.2, Firmware 1.1.2 & Conf Revision 1.0.8.00FD

Although you should really let an automatic download send you the new firmware by placing your MAC on Zooms' web site. (www.zoom.com/ata_update) Don't forget to enable the Automatic Configuration (SIP) update on the Firmware and Configuration Updates Settings screen.



LINKING TWO ZOOM 5801's - HOT LINE - EXTEND A PHONE / FAX LINE WITHOUT A SERVER

Yes it works. Two units back to back without a server will call each other. It does not matter what you dial the other unit will ring. In the past I had to use two Cisco 186’s with two FX-200’s (FXO to FXS converters) for back to back operation. Since the 186 had two ports, one box could be used for two channels in either direction. But if you wanted to move a single line over the net this was an expensive solution (2x186 + 1xFX-200) you may as well pay the call forward fees.

With two Sipuras’ SPA-3000’s back to back, you had two channels a bit less costly than the 186 solution, but only one FXO on each side. A cheaper solution if you needed to link phone systems like the ATT Partners. If however you needed to move more than one line to the other side you had to go back to the Cisco solution or you could buy multiple Sipura units.

Now we have the Zoom. It is similar to using one SPA-3000 on each side. This unit has the ability to be a single or dual channel transfer with FXS and FXO on either side. As an example, we have a pots line we must extend from a remote office to the main office. Placing a unit on the remote side with auto answer set and automatic dialing will route the call to the other unit which will automatically ring a phone or fax attached to it. If you use this as a fax extension, make sure you set the fax pass through mode. I’m using ULaw for faxing pass through testing.

There is one ‘feature missing’ from this scenario. If you bridge a new location – like a newly acquired office – and the internet is down, the call does not automatically fail over to the phone (FXS) port. In the above scenario, I extended a fax line to a remote fax server but if the net is down the Zoom keeps answering the phone and it is not rerouted to the local port for a local fax machine or backup method. I found no way in software to stop this auto answer or force a fail over if the net connection was down (Although pulling the power cord did work:)). I guess at this price you shouldn’t expect it all!

The interesting thing I found is that this unit can be used as a two channel link under certain conditions. I bridged PSTN to VOIP on both units, had both units dial each other (hot line dialing – I used 411 as a dial string) and had a POTS line and a local phone attached to each unit. Using a third and fourth POTS line to test with, dial each Zoom’s POTS connection. Each unit will answer and ring the phone attached to the other unit at the same time. So two way POTS transfer is possible but certain exceptions exist.

These exceptions are basically the same as trying to use both ports on the 5801 with the Asterisk server however it really does not get in the way if you’re bridging two phone systems. Let me give you an idea of the testing. We need one pots connection and one analogue station port on each phone system to call the other. Zoom 1’s phone port (FXS) goes to the input (FXO) connection of phone system 1 and Zoom 1’s telco port (FXO) goes to an analogue extension. Zoom 2’s connections are the same on the second phone system.

Calling the extension that the Zoom’s FXO port is connected to will make the Zoom auto answer and then ring the FXS port of Zoom 2 which is connected to phone system 2’s FXO port so it acts like an incoming call on the second system. Likewise the second system can call the first by also dialing an extension. The POTS line used on both phone systems needs to be isolated in programming so any extensions wanting an outside line on the PBX can not grab the line with the Zoom – it is reserved for incoming calls only from the other unit. (Hence the exception – since if a user on the PBX dials 9 and gets the Zoom line it will bridge to the other phone system just like they dialed the bridging extension)

Now one problem: I could not do complete testing on a live phone system since I was trying to use two older ATT Partner systems, and dialing an internal extension has a special ring cadence. Actually, there is more than one ring pattern – one for internal and one for external calls (others patterns exist like the door ring). The 5801 has the ability to have its ring cadence modified, but I have not yet found the correct sequence for the two ring patterns on the Partner modules.


This page will no longer be updated. To see any additional comments, please see the site located at http://www.siliconvp.us

Paul Norris
Silicon Valley Products


-=
Important notice to product manufacturers, resellers and other people entering new links to this page
=-
If you want to add your company's links, please read the Posting Guidelines for Promoting Products and Services

Where to buy

  • Amazon.com - They always have it all - but your on your own






This unit answers a market need for ATA 911 compliance - to a degree. This ATA has a POTS (FXO) connection in addition to the phone connection (FXS). This POTS connection can be engaged by the dialplan set in the ATA, a loss of the VOIP server as well as dialing a special code. So if I deploy these, based on the config file in the ATA, I can set the 911 to use either the POTS line, the VOIP line or the pots as a backup to the voip line and I believe the VOIP line in backup to the POTS line for emergency numbers.

Having an FXO port makes you want to use it for Asterisk! Well it can to a point. You can have the ATA act as a single line SIP FXO port (in and out) or an ATA, but not concurrently - at this time. Read on.


FIRST LETS GET THE ATA TO WORK ON ASTERISK

Using IE (carefull of IEv7 & Firefox - problems reported), open the web interface to the unit. In VOIP Accounts, set up then 'phone number' and 'auth user name' to the extension you wish to assign in Asterisk. Set up the 'Auth Password' the the password for this extension. Next set the Sip server, Outbound Proxy and Register Domain to the Asterisk server's IP address (or FDDN). I will not go into STUN or NAT so testing should be local.

Set up the extension in Asterisk and you should be able to receive calls to the ATA. Placing calls may mean you have to modify the dial plan as well as a few other paramters.

I set the mode to pass all * and # sequences to the Asterisk server (Subscription Services - Dialing Parameters). This is only overridden by the dial plan template. In addition, I had to remove foreign dial plan codes out of my dial template such as 999, 100 and 11x - especially 100 since I was trying to dial extension 100! (Subscription Services - Emergency Services)


PLACING A CALL OUT THE POTS LINE

First make sure you can call the ATA extension phone from Asterisk and call out Asterisk using the extension phone on the ATA.

The Asterisk server can place a call out the POTS line on the unit if it is set up to: Bridge from VOIP to PSTN, Auto answer, accept any call, bridge anonymous call and disable password checking (Subscription Services)

Ceate a CUSTOM Trunk in Asterisk with the String SIP/xxx where xxx is the extension the ata is set up as - exclude the dial string (OUTNUM) since it does nothing.

This does require a custom dial plan since any trunk or outbound route defined in AMP, TrixBox or AAH will not pass the digits to be dialed properly. Basically there are two ways around this. One is custom dial code and the other is to just engage the outbound line immediately.

To engage the outbound line immediately, just define the cutom trunk as above and an outbound route. Make the outbound route have something like 6| without anything else. Also change your phones mac config file to timeout immediately after pressing the 6. Pressing 6 will execute the trunk and cause the 5801 to answer and go off hook on the pots line. You can then dial the number.

The other option is to create custom code so the normal dial system will work. This is what I did.

Since I am using custom code, I can just add the trunk as another trunk in an already configured outbound route - or for testing just create an ordinary route using the Zoom custom trunk. I edited the AMP file extensions.conf which WILL CHANGE on any AMP update. (But not on every AMP configuration change you make as will extensions_additional.conf). I just made a very small change and it is easy to watch out for on AMP updates. Basically, the dial command allows an additional parameter which is not used by AMP or the custom trunks in AMP and that is to dial additional digits after connection. It is the 'D' Trunk Option.

My custom code is to add 'D(w${OUTNUM})' to the end of the custom dial string (just before the last close brace) in extensions.conf as so:

exten => s,n(skipoutnum),Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS}D(w${OUTNUM}))

The w is a delay to wait for the bridge to complete. This is the quickest fix I could find.

Observations about this configuration:
>>Once the bridge is complete and a call is made from the VOIP server, the analogue phone attached to the ATA can pick up and dial an extension on the VOIP server as a regular extension (if the hot dial string is not engaged as stated in my second configuration below)

>>If the phone attached to the ATA is busy on a VOIP call, any attempt to use the ATA as an outbound trunk causes a call waiting beep to be heard on the phone before the bridge is performed.


RECEIVING AN OUTSIDE CALL USING THE POTS LINE

The VOIP server can receive a call as if it was an external SIP FXO adapter if it is set up to: Bridge from PSTN to VOIP, Auto Answer, Accept any call, accept anonymous and disable passwords. In addition, hot line dialing is engaged with a hot dial string (7777 in AMP) to force the server to go into an inbound trunk mode.


This works great except:

>>If there is a phone attached to the unit and it is picked up it also dials the hot line number, making it an outside call (due to the forced dialed digits) instead of an extension. Trying to avoid the hot dial by making a warm dial (user defined delay, currently 4 sec) would work but the same delay is presented to an outside caller. This gives an outside caller a dial tone for 4 seconds and a 'from-internal' context. So forcing the dial is the easiest way without modifying the context or peer type (friend, user, peer) to avoid misuse.

>>When the FXO picks up, the callerid passed to Asterisk is the phone extension rather than the outside caller. The documentation implies the FXO ID is passed, but the SIP packets say otherwise. I'm still checking on this.


MY OPINION

This unit does have a few problems at this time. The sip status always shows the lag at 60ms or greater when it is 5 feet away and on the same switch, but this does not seem to effect echo or performance. A few bugs do exist and the beta version is addressing these. Someone at Zoom has a sence of humor. I got a response to my sip command that stated it was a COFFEE POT not a sip server. :-o) LOL!!!

I tried the beta 4 software for a few hours but I could not, using the same setup, get it to work. Sip status always showed it as LAGGED, making its use impossible. So the beta is not ready yet, but I may have another crack at it.

It is nice to be able to use a single device for both an ATA and an external single port FXO. Less inventory to buy.

Forget about using both channels with Asterisk at the same time FXS & FXO. At this time there is no convenient way. I have emailed Zoom about allowing this. My idea to maintain 911 compliance on the fxs port is when a 911 or pattern match emergency call is being made is to interrupt the FXO port for a period of time forcing the 911 out the pots line. This would revert back to the VOIP server after a preset time or if a special key combination was entered.

In addition, they should stop the local phone from ringing when the bridge from VOIP to PSTN is engaged. Also the beep should be eliminated if the local phone is off hook. However I do realize that my testing is using the second appearance of a single line so I may just have to wait for a new offering from them.

The deployment tools for large installs that allow you to upload new configurations to your clients also need a little more documentation. They state Perl & PHP must be installed, but minimumn versions are not specified. I'm using PHP 4.3 (I Know its a little behind) and I could not run the utilities properly. Also creating the recommended Apache links didn't work, too much depends on the exact structure, so move the whole thing into the html folder or start coding.

Overall, I like it. I'm going to pick up a few more and deploy them in test areas. Now if I could just get a better firmware...



This was based on Rom 1.0.2, Firmware 1.12 & Conf Revision 1.1.1.01

Also confirmed working ROM 1.0.2, Firmware 1.0.7 & Conf Revsion 1.0.8-00SP

Also confirmed working ROM 1.0.2, Firmware 1.1.2 & Conf Revision 1.0.8.00FD

Although you should really let an automatic download send you the new firmware by placing your MAC on Zooms' web site. (www.zoom.com/ata_update) Don't forget to enable the Automatic Configuration (SIP) update on the Firmware and Configuration Updates Settings screen.



LINKING TWO ZOOM 5801's - HOT LINE - EXTEND A PHONE / FAX LINE WITHOUT A SERVER

Yes it works. Two units back to back without a server will call each other. It does not matter what you dial the other unit will ring. In the past I had to use two Cisco 186’s with two FX-200’s (FXO to FXS converters) for back to back operation. Since the 186 had two ports, one box could be used for two channels in either direction. But if you wanted to move a single line over the net this was an expensive solution (2x186 + 1xFX-200) you may as well pay the call forward fees.

With two Sipuras’ SPA-3000’s back to back, you had two channels a bit less costly than the 186 solution, but only one FXO on each side. A cheaper solution if you needed to link phone systems like the ATT Partners. If however you needed to move more than one line to the other side you had to go back to the Cisco solution or you could buy multiple Sipura units.

Now we have the Zoom. It is similar to using one SPA-3000 on each side. This unit has the ability to be a single or dual channel transfer with FXS and FXO on either side. As an example, we have a pots line we must extend from a remote office to the main office. Placing a unit on the remote side with auto answer set and automatic dialing will route the call to the other unit which will automatically ring a phone or fax attached to it. If you use this as a fax extension, make sure you set the fax pass through mode. I’m using ULaw for faxing pass through testing.

There is one ‘feature missing’ from this scenario. If you bridge a new location – like a newly acquired office – and the internet is down, the call does not automatically fail over to the phone (FXS) port. In the above scenario, I extended a fax line to a remote fax server but if the net is down the Zoom keeps answering the phone and it is not rerouted to the local port for a local fax machine or backup method. I found no way in software to stop this auto answer or force a fail over if the net connection was down (Although pulling the power cord did work:)). I guess at this price you shouldn’t expect it all!

The interesting thing I found is that this unit can be used as a two channel link under certain conditions. I bridged PSTN to VOIP on both units, had both units dial each other (hot line dialing – I used 411 as a dial string) and had a POTS line and a local phone attached to each unit. Using a third and fourth POTS line to test with, dial each Zoom’s POTS connection. Each unit will answer and ring the phone attached to the other unit at the same time. So two way POTS transfer is possible but certain exceptions exist.

These exceptions are basically the same as trying to use both ports on the 5801 with the Asterisk server however it really does not get in the way if you’re bridging two phone systems. Let me give you an idea of the testing. We need one pots connection and one analogue station port on each phone system to call the other. Zoom 1’s phone port (FXS) goes to the input (FXO) connection of phone system 1 and Zoom 1’s telco port (FXO) goes to an analogue extension. Zoom 2’s connections are the same on the second phone system.

Calling the extension that the Zoom’s FXO port is connected to will make the Zoom auto answer and then ring the FXS port of Zoom 2 which is connected to phone system 2’s FXO port so it acts like an incoming call on the second system. Likewise the second system can call the first by also dialing an extension. The POTS line used on both phone systems needs to be isolated in programming so any extensions wanting an outside line on the PBX can not grab the line with the Zoom – it is reserved for incoming calls only from the other unit. (Hence the exception – since if a user on the PBX dials 9 and gets the Zoom line it will bridge to the other phone system just like they dialed the bridging extension)

Now one problem: I could not do complete testing on a live phone system since I was trying to use two older ATT Partner systems, and dialing an internal extension has a special ring cadence. Actually, there is more than one ring pattern – one for internal and one for external calls (others patterns exist like the door ring). The 5801 has the ability to have its ring cadence modified, but I have not yet found the correct sequence for the two ring patterns on the Partner modules.


This page will no longer be updated. To see any additional comments, please see the site located at http://www.siliconvp.us

Paul Norris
Silicon Valley Products


-=
Important notice to product manufacturers, resellers and other people entering new links to this page
=-
If you want to add your company's links, please read the Posting Guidelines for Promoting Products and Services

Where to buy

  • Amazon.com - They always have it all - but your on your own






Created by: Paul2006, Last modification: Fri 06 of Apr, 2007 (01:07 UTC) by Jbrock1
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